1. Field of the Invention
The present invention relates to communication networks through which compressed data is transmitted. In particular, an integrated digital network employing mixed compression modes for speech is disclosed.
2. Description of Related Art
Historically, speech compression in communication networks has been available on a point-to-point basis only. As illustrated in FIG. 1, a call from telephone 1 enters a network at point A, and is passed through speech processing equipment 2 at point A. The call is then transmitted in a compressed mode across a link 3, to point B, where conjugate speech processing equipment 4 decodes the compressed data to recover the original signals and supplies them in a mode compatible with the receiving station 5.
More recently, as illustrated in FIG. 2, equipment has been available that allows end-to-end speech compression on a more complex, multi-hop circuit. For instance, such networks are provided by multiple integrated digital network exchange switches such as those provided by Network Equipment Technologies, Inc., the assignee of the present invention. Such networks, for example, allow speech signals entering the network from a first station 6, to enter speech processing equipment 7 at point A, and to be passed in a compressed mode through link 8 to point X, where equipment 9 transmits the compressed data on link 10 to point Y. At point Y, equipment 11 transmits the compressed data on link 12 to point Z, where equipment 13 transmits the compressed data on link 14 to point B. At point B, conjugate speech expansion equipment 15 reconstitutes the original speech signal which is then delivered to the receiving station 16.
One example of a prior art end-to-end speech compression algorithm is illustrated in U.S. Pat. No. 4,679,187 entitled ADAPTIVE TRUNK-COMPRESSION SYSTEM WITH CONSTANT GRADE OF SERVICE; invented by David R. Irvin. In the Irvin patent, varying traffic intensity is handled with a dynamic trunk compression system for end-to-end processing of a speech signal entering the network. The adaptive compression system of Irvin involves varying the sample rate for pulse code modulation (PCM) of a speech signal in order to vary the number of channels that can be carried on a single trunk in the system. The sample rate and number of channels on the trunk are calculated upon coding of the speech signal and those parameters are transmitted with the speech signal to the receiving station. At the receiving station the parameters are read and the speech decoded according to the same dynamic compression algorithm. Irvin addresses the problem of making efficient use of transmission resources by adapting the coding rate for PCM speech signals. However, as an end-to-end system, it illustrates the problem addressed by the present invention. That is, all receiving stations must have equipment capable of decoding the data from the adaptive PCM formats.
In the end-to-end networks of the prior art such as taught by Irvin, a single compression algorithm is used to assure complete compatibility throughout the network for transmitting and receiving signals. As the traffic increases within a network of the prior art, either the whole network requires modification to a compression algorithm to provide higher data flow in existing links, or additional switches and links must be purchased and added to the system. Expansion of a network to handle increased traffic load can be an expensive, complicated task.